Re: [asterisk-users] IAX and Variables
I thunk so Let me see -Original Message- From: Mikhail Lischuk Sender: asterisk-users-bounces@lists.digium.comDate: Tue, 08 Oct 2013 01:08:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX and Variables -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX and Variables
Wont this work? exten => 18,1,Set(CDR(accountcode)=${IAXVAR(ACCOUNTID)}) accountcode is not read-only property so it should be writeable. Phibee Network Operation Center писал 07.10.2013 21:05: > Hi > > a new small question ;=) > > We have two Asterisk, connected in IAX2. > > On the first, in dialplan, we have: > exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) > we sent into the IAXVAR "ACCOUNTID" the accountcode. > > On the second, in dialplan, we have: > exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) > > That's work, the second server get the variable. > > I want now said at the second server that accountcode = > ${IAXVAR(ACCOUNTID)}, > for use this accoundcode in CDR. On second server, in cdr_mysql.conf i have: > > [columns] > alias start => calldate > alias end => callend > alias clid => clid > alias src => src > alias dst => dst > alias dcontext => dcontext > alias channel => channel > alias dstchannel => dstchannel > alias lastapp => lastapp > alias lastdata => lastdata > alias duration => duration > alias billsec => billsec > alias disposition => disposition > alias amaflags => amaflags > alias accountcode => accountcode > alias userfield => userfield > alias uniqueid => uniqueid > > But where i can put into the config that for this cdr entry accountcode > = ${IAXVAR(ACCOUNTID)} ? > > thanks for your help > > jerome -- With Best Regards Mikhail Lischuk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX and Variables
Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I want now said at the second server that accountcode = ${IAXVAR(ACCOUNTID)}, for use this accoundcode in CDR. On second server, in cdr_mysql.conf i have: [columns] alias start => calldate alias end => callend alias clid => clid alias src => src alias dst => dst alias dcontext => dcontext alias channel => channel alias dstchannel => dstchannel alias lastapp => lastapp alias lastdata => lastdata alias duration => duration alias billsec => billsec alias disposition => disposition alias amaflags => amaflags alias accountcode => accountcode alias userfield => userfield alias uniqueid => uniqueid But where i can put into the config that for this cdr entry accountcode = ${IAXVAR(ACCOUNTID)} ? thanks for your help jerome -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones flashing but not ringing
> Have you tried restarting the phone instead of Asterisk? I don't think that > Asterisk sends > separate commands to the bell and to the call LED. Since the LED is flashing, > it is likely that > the "SIP INVITE" signal from Asterisk is ok. Also the ring tone normally does > not come from > Asterisk itself. > We tried it and it doesn't help. It's not one phone, multiple phones do it at the same time. I think it's related to Asterisk SLA - maybe device states get messed up. Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADSL and VPN router
If you are looking for software i would go for pfsense. if your are looking for hardware, i would go for mikrotik. Hello; I am looking for ADSL that supporting VPN so we can connect to it from our IPhone using the VPN to be able to register at the asterisk PBX. Any recommended one that is doing fine with voice? Also, does it support bandwidth priority or shaping for the protocols? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple resetcdr calls have no effect
To answer my question, set unanswered=yes in cdr.conf Source: http://lists.digium.com/pipermail/asterisk-users/2009-December/241749.ht ml From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Monday, October 07, 2013 7:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple resetcdr calls have no effect Hi All Using Asterisk 11. My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n, hangup() I expected at least 3 CDR's in MySQL. I see only one. Tried playing with cdr.conf settings but apparently they have no effect. The cdr_mysql.conf is pretty generic. *CLI> cdr show status Call Detail Record (CDR) settings -- Logging: Enabled Mode: Batch Log unanswered calls: No Log congestion: No * Batch Mode Settings --- Safe shutdown: Enabled Threading model: Scheduler only Current batch size: 0 records Maximum batch size: 3 records Maximum batch time: 1 second Next batch processing time: 0 seconds * Registered Backends --- mysql Adaptive ODBC csv cdr-custom I checked /var/log/asterisk/cdr-csv/Master.csv. There is only one record for the call. Expected the resetcdr to be synchronous. Any help would be gratefully acknowledged. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring Busy ?
Hi I use asterisk with Realtime/Mysql. I have put a Call-limit at 1, but if the SIP account receive two call in same time, the second call don't ring busy. Ayone know a solution for the second call get a busy ring ? Best Regards Jerome -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AppKonference 2.4
Hi, I have released AppKonference 2.4 today. This release includes a frame cache for asterisk frames which should be more efficient than the asterisk frame cache is for this type of application. It's ulaw/alaw only for now. This release also added a speaker list which the conference thread uses to mix speaking members. This should be more efficient for large conferences when relatively few members are speaking. -- Paul Albrecht -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones flashing but not ringing
Have you tried restarting the phone instead of Asterisk? I don't think that Asterisk sends separate commands to the bell and to the call LED. Since the LED is flashing, it is likely that the "SIP INVITE" signal from Asterisk is ok. Also the ring tone normally does not come from Asterisk itself. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi not detecting hangup when analog forwarding
After posting her, I've found this thread http://forums.asterisk.org/viewtopic.php?f=1&t=83088 In it I read: " The normal PSTN behaviour is calling party clearing, with a timeout of 2 to 6 minutes if the called party is the only one that clears." Can someone explain a bit further why "calling party clearing" keeps "analog forwarding from working" ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones flashing but not ringing
We have been using Asterisk SLA for a while with Cisco SPA series phones. Once in a while the phones flash, but not ring when a call comes in. We can pick it up and talk to the caller even if that's the case. This is pretty random (might not happen for couple of weeks). The quick solution is to restart Asterisk which we are trying to avoid. What might cause this? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi not detecting hangup when analog forwarding
Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Asterisk2 comes in, it is correctly entering the above remote context and an extension on Asterisk2 receives an incoming call: bothe Asterisk extension are talking to each other and everything run fine (two ways audio) except that hangup is not detected by Asterisk1 (no matter which Asterisk2 extension is hanging up): - Asterisk2 shows non ongoing call, - Asterisk1 show 2 living channels: I need to kill one of them to restore expected status. When I'm using another dialplan with which an incoming call is passed to a local extension, hangup is correctly detected. Am I trying to do something that can't be done (forwarding from one line to another) ? Any clue ? Suggestions. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple resetcdr calls have no effect
Hi All Using Asterisk 11. My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n, hangup() I expected at least 3 CDR's in MySQL. I see only one. Tried playing with cdr.conf settings but apparently they have no effect. The cdr_mysql.conf is pretty generic. *CLI> cdr show status Call Detail Record (CDR) settings -- Logging: Enabled Mode: Batch Log unanswered calls: No Log congestion: No * Batch Mode Settings --- Safe shutdown: Enabled Threading model: Scheduler only Current batch size: 0 records Maximum batch size: 3 records Maximum batch time: 1 second Next batch processing time: 0 seconds * Registered Backends --- mysql Adaptive ODBC csv cdr-custom I checked /var/log/asterisk/cdr-csv/Master.csv. There is only one record for the call. Expected the resetcdr to be synchronous. Any help would be gratefully acknowledged. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
Darryl Moore wrote: Thank you Steve, and I read a bit more on the web on this subject including your own well reasoned page at http://www.soft-switch.org/patents/index.html However, despite wide acceptance of the patentability of such codecs (unfortunately), whether they are in fact software patents or not appears to be a matter of opinion. The FSF and Fedora both refer to codec patents as being software patents. http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/ http://fedoraproject.org/wiki/Software_Patents A quick google search of both terms will show that there are a great many people who see codec patents as software patents, so I don't think I am alone there. Law is ALWAYS open to interpretation, so that is not surprising. See if you can get any lawyer, and especially a patent attorney, to give you a definitive answer! You will not get one. Seldom will you ever get an "eggspurt legal opinion" Any good lawyer will tell you "maybe", or if there is any doubt don't do it! Law is not precisely measurable. No meter or O'scope to assist here. Any A**hole can sue anyone for the filing fee, and the results are up to the opinion of a judge or jury. The lawyers want it that way, so it isn't ever going to be any different. John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi incoming call detection and hangup detection durations.
Hi, I've set an Asterisk 11 box with a TDM400 board and Dahdi 2.7.0.1. I've connected an FXS port to an FXO one and issued a couple of channel originate command to measure the duration Asterisk/Dahdi needs to detect a dahdi call is coming in. Basically, using EPOCH variable, I'm reading a 2 or 3s duration with the followinf AEL2 dialplan: context remote { s => { if ("x${DB(Start/FXS1)}" != "x") { Duration=$[${EPOCH} - ${DB(Start/FXS1)}]; Verbose(0,Duration is ${Duration}); } Answer(); Wait(5); HangUp(); }; }; context mylocal { 1 => { DB(Start/FXS1)="${EPOCH}"; Dial(DAHDI/1); HangUp(); }; }; How should I rate this 2s or 3s duration ? Can I shorten this value ? On the opposite, which settings would significantly increase this duration ? With the same king of dialplan, I observed hangup needed 4 or 5s to propagate from one port to the other. How should I rate this duration ? Can I also shorten this value ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADSL and VPN router
Hello; I am looking for ADSL that supporting VPN so we can connect to it from our IPhone using the VPN to be able to register at the asterisk PBX. Any recommended one that is doing fine with voice? Also, does it support bandwidth priority or shaping for the protocols? Regards Bilal-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users